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User Manual
GXP-2000
Enterprise IP Phone
For Firmware Version 1.1.1.14
Grandstream Networks, Inc.
www.grandstream.com
Table of Contents
1. Welcome.........................................................................................................................4
2. Installation......................................................................................................................5
2.1 What is Included in the Package ..........................................................................5
2.2 Connecting Your Phone .......................................................................................5
2.3 Wall Mount...........................................................................................................6
2.4 GXP2000 Extension Board ..................................................................................7
2.5 Safety Compliances..............................................................................................9
2.6 Warranty ...............................................................................................................9
3 Product Overview..........................................................................................................11
3.1 Key Features.......................................................................................................12
3.2 Hardware Specification ......................................................................................13
4 Using GXP-2000 IP Phone ...........................................................................................15
4.1 Getting Familiar with LCD ....................................................................................15
4.2 Getting Familiar with Keypad............................................................................16
4.3 GUI Menu Chart.................................................................................................18
4.4 Making and Answering Phone Calls..................................................................18
4.4.1 Handset, Speakerphone and Headset Mode ................................................18
4.4.2 Multiple SIP Accounts and Lines................................................................18
4.4.3 Making Calls ...............................................................................................19
4.4.4 Making Calls using IP Address ...................................................................19
4.4.5 Receiving Calls............................................................................................20
4.4.6 Call Hold......................................................................................................20
4.4.7 Call Waiting and Switch between Calls ......................................................20
4.4.8 Call Transfer ................................................................................................20
4.4.9 3-Way Conferencing ...................................................................................21
4.4.10 Checking Message and Message Waiting Indication ..............................21
4.4.11 Mute and Delete .......................................................................................21
4.4.12 Multi-functional Button/Extensional board .............................................22
4.4.12.1 Speed Dial ..............................................................................................22
4.4.12.2 Asterisk Busy Line Field .......................................................................22
4.4.12.3 Extension Board.....................................................................................22
The fifty-six Multi-functional buttons on Extension board function the same as
Multi-functional buttons on the phone base station, except that they can not be
Line keys. .............................................................................................................22
4.4.13 Page Dialing .............................................................................................22
4.5 Call Features.......................................................................................................22
4.6 Customized idle screen.......................................................................................23
4.7 XML phone book downloading .........................................................................23
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5 Configuration Guide......................................................................................................24
5.1 Configuration with Keypad....................................................................................24
5.2 Configuration with Web Browser ......................................................................27
5.2.1 Access the Web Configuration Menu .........................................................27
5.2.2 End User Configuration...............................................................................27
5.2.3 Advanced User Configuration.....................................................................35
5.2.4 Saving the Configuration Changes ..............................................................49
5.2.5 Rebooting the Phone from Remote .............................................................50
5.3 Configuration through Central Provisioning Server...........................................50
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Firmware Upgrade ...................................................................................................52
6.1 Firmware Upgrade through TFTP/HTTP ..............................................................52
6.2 Configuration File Download.............................................................................53
6.3 Firmware and Configuration File Prefix and Postfix.........................................53
6.4 Managing Firmware and Configuration File Download....................................53
7 Restore Factory Default Setting..................................................................................54
Appendix I Glossary of Terms.....................................................................................55
Appendix II GUI Menu Chart........................................................................................62
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1. Welcome
Thank you for purchasing Grandstream award-winning GXP-2000 Enterprise IP Phone.
You made an excellent choice and we hope you will enjoy all its capabilities.
Grandstream's award-wining GXP-2000 SIP IP phone is the innovative enterprise IP
telephone that offers a rich set of functionality and superb sound quality. They are fully
compatible with SIP industry standard and can interoperate with many other SIP
compliant devices and software on the market.
Grandstream GXP-2000 has been awarded the Best of Show product in 2005 Internet
Telephony Conference and Expo.
This document is subject to changes without notice. The latest electronic version of this
user manual is available for download from the following location:
http://www.grandstream.com/user_manuals/GXP2000.pdf
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2. Installation
2.1 What is Included in the Package
The GXP-2000 phone package contains:
1)
2)
3)
4)
5)
One GXP-2000 Main Case
One Handset
One Phone Cord
One Universal Power Adaptor
One Ethernet Cable
2.2 Connecting Your Phone
Following is a backside picture of GXP-2000; each connection port is labeled with the
name in the following table:
EXT
PC
LAN+PoE
POWER
HEADSE
T
The table below describes the connectors on the GXP-2000 phone:
5
EXT
Extension connection for extended keypad(will be
implemented in the future)
LAN/PoE
10/100 Switch LAN port for connecting to Ethernet.
Support PoE (802.3af). Draws power from both spare line
and signal line
PC
POWER
HEADSET
10/100 Switch port for connecting PC
5V power port
3.5mm Headset port
2.3 Wall Mount
GXP-2000 can be wall mounted. There are two wall mount holes on the bottom of the
GXP-2000 main body:
Top Wall
Mount hole
Bottom Wall
Mount hole
User can simply place the device against the wall with two holes placed to the fixed
hanger position on the wall.
Handset
Rest
Tab
Tab with
extension down
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Tab with
extension up
After wall mounting the main body of GXP-2000, user will need to pull out the tab
(extension downward) from handset cradle on the top of the handset rest, and rotate the
tab and plug it into the slot with the extension up for handset holding.
2.4 GXP2000 Extension Board
GXP-2000 supports up to 2 extension boards from firmware 1.1.1.1. GXP2000
Extension Board expands more Multi Purpose Keys for advanced business phone
applications. One GXP2000 Extension Board carries 56 Multi Purpose Keys. Simply
connect the extension board with GXP2000 via PS2 cable in the same package, the
GXP2000 will automatically reboot itself and initiate the extension board.
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The GXP2000 will boot up first, and all LED indicators on GXP2000 will be solid on
for a while; the status light at the top right corner of the extension board will blink in
red; and then all of the LED indicators on the extension board will flash three times; and
then the status light at the top right corner of the extension board will turn to solid
green.
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Note: If you are using GXP2000 with HW0.4, power supply in the same package should
be plugged in the extension board. Hardware version can be found in the GUI STATUS
Menu (you can use the down arrow key to browse GUI menu when the phone is onhook).
2.5 Safety Compliances
The GXP-2000 phone is compliant with various safety standards including FCC/CE. Its
power adaptor is compliant with UL standard. The phone should only be operated with
the universal power adaptor provided with the package. Damages to the phone caused
by using other unsupported power adaptors are not covered by the manufacturer’s
warranty.
2.6 Warranty
Grandstream has a reseller agreement with our reseller customer. End user should
contact the company from whom you purchased the product for replacement, repair or
refund.
If you purchased the product directly from Grandstream, contact your Grandstream
Sales and Service Representative for a RMA (Return Materials Authorization) number.
Grandstream reserves the right to remedy warranty policy without prior notification.
Warning: Please do not attempt to use a different power adaptor. Using
other power adaptor may damage the GXP-2000 and will void the
manufacturer warranty.
Caution: Changes or modifications to this product not expressly approved by
Grandstream, or operation of this product in any way other than as detailed by this
User Manual, could void your manufacturer warranty.
9
Information in this document is subject to change without notice. No part of this document may be
reproduced or transmitted in any form or by any means, electronic or mechanical, for any purpose without
the express written permission of Grandstream Networks, Inc.
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3 Product Overview
GXP-2000 series IP phone is designed to be an enterprise telephone, which could also
be used in residential household. The following photo illustrates the appearance of a
GXP-2000 IP phone.
Front View
Back View
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3.1 Key Features
Grandstream GXP-2000 IP Phone is a next generation enterprise IP telephone based on industry open
standard SIP (Session Initiation Protocol). Built on innovative technology, Grandstream IP Phone
features market leading superb sound quality and rich functionalities at mass-affordable price.
Software Feature:
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
Support SIP 2.0, TCP/UDP/IP, PPPoE, RTP/RTCP, SRTP by SDES, HTTP, ARP/RARP,
ICMP, DNS, DHCP, NTP, TFTP, STUN, etc.
Support up to 4 SIP accounts and up to 11 media channels.
The two Ethernet ports can be configured to function as NAT router.
Powerful digital signal processing (DSP) to ensure superb audio quality; advanced adaptive
jitter control and packet loss concealment technology
Support popular codecs including G711 (a-law and u-law), G.723.1 (6.3K), G.726 (32K),
G.729A/B, and GSM.
Support Caller ID/name display or block, Call waiting caller ID, Hold, Call Waiting, Call
Transfer(consultative/blind), 3-way conference, Call Forward, in-band and out-of-band DTMF,
Do-not-Disturb, SIMPLE/PRESENSE, phone book downloading, customized idle screen,
Asterisk BLF, speed dial, paging, message waiting indicator, etc.
Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise
Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain Control).
Support standard encryption and authentication (DIGEST using MD5 and MD5-sess).
Support for Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS).
Support automated NAT traversal without manual manipulation of firewall/NAT.
Provide easy configuration through manual operation (phone keypad), Web interface or
automated provisioning by downloading encrypted configuration file via HTTP/TFTP for mass
deployment
Support firmware upgrade via TFTP or HTTP.
Support customized configuration/firmware file name by attaching prefix and post fix.
Support GXP-2000 Extension board for multi-purpose functionality.
Support phonebook downloading via HTTP/TFTP.
Support customizable idle screen by downloading XML format file via HTTP/TFTP
Support SIP MESSAGE method (RFC 3428); stores up to 100 incoming IM messages.
Support SIP PUBLISH method (RFC 3903), SIP Presence package (RFC 3856, 3863) for use
of 7 MFKs and GXP-2000EXT, SIP Dialog package (RFC 4235)
Support Power over Ethernet (PoE) IEEE standard 802.3af
Support Headset which will auto switch to Headset when plugged in
Support 10/100 Full/Half Duplex Ethernet Switch with LAN and PC port, Ethernet polarity can
be auto detected, thus either straight through or twist cable can be used.
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3.2 Hardware Specification
The table below describes the hardware specification of GXP-2000:
Model
GXP-2000
LAN interface
Power over Ethernet
2xRJ45 10/100Base-T with PoE (802.3af)
IEEE 802.3af standard, can draw power from both spare lines
or signal lines from Ethernet
3.5mm Headset port
11 LED with different light pattern in RED color
Input: 100-240VAC 50-60 Hz
Output: +5VDC, 1200mA,
UL certified
215mm (W)
220mm (D)
57mm (H)
0.82kg (1.8lbs)
40 - 130oF
5 – 45oC
10% - 90%
(non-condensing)
FCC / CE / C-Tick
Headset Jack
LED
Universal Switching
Power Adaptor
Dimension
Weight
Temperature
Humidity
Compliance
The picture below shows the handset and headset connectors’ wiring schema.
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As show in the schema, the left side is pin assignment for a RJ11 interface headset; while the right side
is showing a normal 3.5mm headset plug. A 3.5mm to 2.5mm plug converter is required if user want to
user normal 2.5mm cell phone headset. The plug converter can be purchased from any electronics
department store like Radio Shack.
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4 Using GXP-2000 IP Phone
4.1 Getting Familiar with LCD
GXP-2000 phone has a numeric LCD of 64(rows) x 131 (columns) in pixels. Here is the display when
all segments illuminate:
The LCD is equipped with a backlight. When the phone is configured properly and in the normal idle
state, the backlight is off. Whenever an event occurs, the backlight turns
on automatically and brings the user’s attention.
Icon
LCD Icon Definitions
Network Status Icon:
FLASH in the case of Ethernet link failure
OFF if IP address or SIP server is not found
ON if IP address and SIP server are located
Phone Status Icon:
OFF when the handset is on-hook
ON when the handset is off-hook
Speaker Phone Status Icon:
FLASH when phone rings or a call is pending
OFF when the speakerphone is off
ON when the speakerphone is on
Handset, Speakerphone and Ring Volume Icon:
0-7 scales to adjust handset / speakerphone / ring volume
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Real-time Clock:
Synchronized to Internet time server
Time zone configurable via web browser
AM
PM
Time Icon:
AM for the morning
PM for the afternoon
4.2 Getting Familiar with Keypad
Message Waiting
Indicator
Line 1-4 Keys
Menu Keys
Speed Dial /
Configurable
line indicators
Mute/Delete
Message
Conference
Transfer
RJ11
Hold Speaker
Send/Re-Dial
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Standard Keypad
GXP-2000 phone has 35 key buttons:
Key Button
Key Button Definitions
LINE1-LINE4
4 Line keys with LED, can be extended to 11 Lines with the use of
7 Speed Dial Keys on the right
MULTI-FUNCTION
7 Multi-functional keys with LED that can be configured to use for
speed dial/Asterisk BLF/presence watcher
UP ↑
Scroll up Menu item when phone is in MENU mode
Or increase handset/speakerphone volume when phone is ACTIVE
Or access the missed calls menu when phone is in IDLE mode
DOWN ↓
Scroll down Menu item when phone is in MENU mode
Or reduce handset/speakerphone volume when phone is ACTIVE
Or access the Phone Book when phone is in IDLE mode
LEFT Å
Shift cursor to left
RIGHT Æ
Shift cursor to right
MENU ●
Enter MENU mode when phone is in IDLE mode.
It is also the ENTER key once entering MENU
TRNF
Transfer an ACTIVE call to another number
CONF
Bring Calling/Called party into conference
MSG
Enter to retrieve voice mails or other messages
MUTE/DEL
HOLD
SPEAKER
SEND
0 - 9, *, #
Mute an ACTIVE call; or Delete a key entry, call log, voice mail
and etc
Or use of MUTE/DEL key during incoming call ringing state to
reject call using SIP 486 message
Or act as toggle key to turn DND on and off during idle
Temporarily hold an ACTIVE call
Enter hands-free mode
Dial a new number or Redial the last number dialed. After entering
the phone number, pressing this key would force a call to go out
immediately before timeout
12 standard Digit, * and # keys are usually used to make phone
calls
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4.3 GUI Menu Chart
Please see the Appendix II.
4.4 Making and Answering Phone Calls
4.4.1 Handset, Speakerphone and Headset Mode
The regular Handset mode can be switched with either the Speaker mode (Hand free) or the Headset
mode, however, whenever the Headset is plugged in, Speaker mode will be switched to the Headset
mode automatically.
To Switch between Handset and Speaker/Headset, simply press the Hook Flash in the Handset cradle
or the Speaker button.
4.4.2 Multiple SIP Accounts and Lines
GXP-2000 can support up to 4 independent SIP accounts. Each account is capable of independent SIP
Server, user and NAT settings among others. GXP-2000 supports up to 11 concurrent audio channels
arbitrarily assigned to these SIP accounts -- they can be used in any combination as long as the server
allows it. Speed dial numbers configured must be associated to a specific SIP account.
Each of the 4 LINE buttons (LINE1-LINE4) is “virtually” mapped to each SIP account. In off hook
state, when user chooses an idle line, the name of the account (as configured in the web interface) will
be displayed in the LCD while a dial tone is being played out. For example, if the 4 SIP accounts are
named FWD, SIPPHONE, BROADVOICE, and PBX respectively and they are all active and
registered. When LINE1 is pressed, user will hear dial tone and see “FWD”. When LINE2 is pressed,
user will hear dial tone and see “SIPPHONE”. When LINE3 is pressed, user will hear dial tone and see
“BROADVOICE”. When LINE4 is pressed, user will hear dial tone and see “PBX”.
For outgoing calls, GXP-2000 will pick up the LINE pressed, which will be lit up in solid red color.
User can switch the dialing account before dialing any digits by pressing the same LINE button one or
more times. If user continues to press one LINE, the selected account will circulate among the
registered accounts. For example, when LINE1 is pressed, LCD displays “FWD”. If LINE1 is pressed
again, LCD displays “SIPPHONE” and the subsequent call will be made through SIP account 2.
For incoming calls, if an account is configured and registered, all incoming calls for that account will
attempt to use its corresponding LINE if it is not in use. When the “virtually” mapped line is in use,
GXP-2000 will flash the next available LINE (from Left to Right, then Top to Bottom) in red color.
LINE 5 to 11 cannot be picked like LINE 1 to 4. This happens automatically. When an incoming call
arrives while all of the 4 LINE (1-4) channels are in use, LINE5 will be selected. When all 4 LINE (14) channels are in use, and user places an active call on hold, user can on-hook and off-hook to activate
the next available channel (LINE5 or whatever the next one). When any one of the 7 functions keys is
associated with a call, they function as LINE keys; otherwise they function as speed dial keys. (So
when LINE 5 is in use, you cannot use speed dial 1, but speed dial 2-7 still work).
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A LINE is defined as “ACTIVE” when it is making or receiving a call, and its corresponding LINE
LED is lit up in solid RED.
4.4.3 Making Calls
There are many ways to make phone calls:
1. Make Handset/SPEAKER/Headset off hook, or press the available LINE key to select a SIP
account, the corresponding LINE LED will light up in solid red. Enter the phone numbers and
press the SEND key.
2. Make Handset/SPEAKER/Headset off hook, or press the available LINE key, the
corresponding LINE LED will light up in solid red. Press the SEND button to redial the last
number called.
3. Make Handset/SPEAKER/Headset off hook, or press the available LINE key, the
corresponding LINE LED will light up in solid red. Press the Speed Dial key to call the preset
calling party number.
4. Press the DOWN button, then select the number in the Phone Book menu you want to call by
pressing the Menu button, and then press the Menu button again to call this number.
5. Press the UP button, then select the number in the Missed Calls you want to call by pressing the
Menu button, and then press the Menu button again to call this number.
Note:
• Once pressed, the dialed number is displayed on the LCD as the corresponding DTMF tone is
played out.
• If the “SEND” button is not pressed after the phone number, the phone will wait for 4 seconds
before initiating the call.
4.4.4 Making Calls using IP Address
Direct IP calling allows two phones to talk to each other in an ad hoc fashion without a SIP proxy.
VoIP calls can be made between two phones if
•
•
•
Both phones have public IP addresses, or
Both phones are on a same LAN using private or public IP addresses, or
Both phones can be connected through a router using public or private IP addresses.
To make a direct IP calling, disable “Use Random Port” option at advanced web configuration page,
and then press “Menu” button, and then select “Direct IP Call” submenu to enter the direct IP call
interface, and then enter the 12-digit target IP address, and then press Menu button twice to make the
call.
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From 1.1.0.13 firmware build, GXP2000 begins to offer Quick IP-call feature: first make
Handset/SPEAKER/Headset off hook, and then press # key and enter the last 3-digits of the target IP
address, and then press the SEND key or # key. To use this feature, you need to enable Quick IP-call
mode in the advanced web configuration page.
4.4.5 Receiving Calls
There are two states when GXP-2000 receives a call:
1. When receiving an initial call. Besides ringing with selected Ring Tone, the corresponding
account LINE will flash in red, taking Handset/SPEAKER/Headset off hook will enable user to
hear the calling party in Handset/SPEAKER/Headset.
2. When receiving second or more incoming calls, besides playing stutter Call Waiting tone,
GXP-2000 will pick up the corresponding account LINE or the next available LINE as
described in section 4.4.2.
4.4.6 Call Hold
While in conversation, pressing the “HOLD” button will put the other party on hold. User can resume
the conversation by pressing the corresponding LINE. User will also automatically put the current line
on “HOLD” by pressing another available LINE for making or receiving other phone calls.
4.4.7 Call Waiting and Switch between Calls
GXP-2000 can support up to 11 Lines, user can switch to another line for making or answering calls
by pressing the line button, and automatically put an ACTIVE call on Hold.
When receiving second or more incoming calls, besides playing a stutter Call Waiting tone, GXP-2000
will pick up the corresponding account or the next available LINE as described in section 4.4.2.
4.4.8 Call Transfer
GXP-2000 supports both BLIND and ATTENDED Transfer:
1. Blind Transfer: When a LINE is “ACTIVE”, user will get a dial tone by pressing the “TRNF”
button, and then dial the number and press the “SEND” button. This will transfer the other
party in the corresponding LINE to the dialed number.
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2. Attended Transfer: When in conversation with an “ACTIVE” LINE as defined in section 4.3.2,
and another LINE that is put on-hold, user can press “TRNF” button, then press the intended
LINE that is on “HOLD”.
If there is no LINE on HOLD, user will need to make a call and thus automatically puts the
current ACTIVE LINE on HOLD.
NOTE:
•
4.4.9
Transferring calls across SIP domains needs to be supported by SIP services.
3-Way Conferencing
GXP-2000 supports 3-way conferencing. With one LINE ACTIVE and another LINE on HOLD, press
the CONF button then the LINE that is on HOLD will join the three parties together in a conference.
If the conference holder wishes to end a conference, simply press HOLD, which breaks the conference
and places both parties on hold. User can then talk to each individual party by selecting the
corresponding LINE.
4.4.10 Checking Message and Message Waiting Indication
When GXP-2000 is on-hook, pressing the MSG button will trigger the phone to call the VM Server
(VMS) configured for the primary account. If a line/account is selected first, it dials the VMS
configured for that account.
The MWI (Message Waiting Indicator) LED will flash in red color in three quarters of a second when
voicemail server sends message waiting information to GXP-2000.
4.4.11 Mute and Delete
When in conversation with an ACTIVE LINE, pressing “MUTE/DEL” will mute the conversation, that
is, you can hear the other party but the other party cannot hear you. Pressing the button again will
resume the conversation.
When dialing a number, pressing “MUTE/DEL” button will delete the last entered digit.
When the phone is in idle status, pressing “MUTE/DEL” button will activate “Do-not-Disturb”
function on the phone. Pressing MUTE/DEL” button again will deactivate DND function.
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4.4.12 Multi-functional Button/Extensional board
4.4.12.1 Speed Dial
The seven Multi-functional buttons can be configured for speed dial function. A vertical rectangle pad
on the keypad is provided to label Speed Dial numbers. Pressing the speed dial button will
automatically initiate a call to the destination extension.
Note:
•
•
When an incoming call arrives while all of the 4 LINE (1-4) channels are in use, the Multifunctional buttons will function as LINE keys, and flash the light for the next incoming
call. User can press the button to pick up the call.
When any one of the 7 functions keys is associated with a call, they function as LINE keys. In
this case, speed dial/BLF function will not work. For ex. when first Multi-functional button is
in use, you cannot use it for speed dial/BLF.
4.4.12.2 Asterisk Busy Line Field
The seven Multi-functional buttons can be configured for Asterisk Busy Line Field function with a
specified account. When Asterisk BLF is configured on one of the Multi-functional buttons, Speed
Dial function on it will still work when it shows idle status.
4.4.12.3 Extension Board
The fifty-six Multi-functional buttons on Extension board function the same as Multi-functional
buttons on the phone base station, except that they can not be Line keys.
4.4.13 Page Dialing
When off-hook, you see "LINEx: DIAL USING" you can press the ROUND button and you will see
"LINEx: PAGE USING" you can toggle between the modes by pressing the button BEFORE any
DTMF digits are dialed. On the called party, "Allow Auto Answer by Call-Info" on advance page
needs to be set to Yes for Paging function. The called party will receive an INVITE with call-info
header that has “answer-after=0", then it will answer the call automatically.
Note: this does not work with Asterisk or other proxies that does not pass along the Call-Info
header. There are workarounds for Asterisk (server side setup).
4.5 Call Features
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GXP-2000 series phone supports a list of call features: Caller ID Block (or Anonymous Call),
Disable/Enable Call Waiting, Call Forward on Busy, Delay, or Unconditional, etc.
Following table shows the call features of GXP-2000 series phone.
Key
*30
*31
*67
*82
*70
*71
*72
*73
*90
*91
*92
*93
Call Features
Block Caller ID (for all subsequent calls)
Send Caller ID (for all subsequent calls)
Block Caller ID (per call)
Send Caller ID (per call)
Disable Call Waiting. (Per Call)
Enable Call Waiting (Per Call)
Unconditional Call Forward
To use this feature, dial “*72” and get the dial tone. Dial the forward
number and “#” for a dial tone, and then hang up.
A call forward icon on status line will be seen when account 1 is set
to unconditional call forward.
Cancel Unconditional Call Forward
To cancel “Unconditional Call Forward”, dial “*73” and get the dial
tone, then hang up.
Busy Call Forward
To use this feature, dial “*90” and get the dial tone. Dial the forward
number and “#” for a dial tone, and then hang up.
Note: Busy forward functions only when all media channels are in
use. Since GXP-2000 supports up to 11 lines, this function will not
Cancel Busy Call Forward
To cancel “Busy Call Forward”, dial “*91” and get the dial tone, then
hang up.
Delayed Call Forward
To use this feature, dial “*92” and get the dial tone. Dial the forward
number and “#” for a dial tone, and then hang up.
Cancel Delayed Call Forward
To cancel this Forward, dial “*93” and get the dial tone, then hang up.
4.6 Customized idle screen
GXP-2000 supports the feature that allows customer to customize the idle LCD screen display. Please
send email to [email protected] for detailed information.
4.7 XML phone book downloading
GXP-2000 supports the feature that allow the user to download XML phone book. Please send email to
[email protected] for detailed information.
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5 Configuration Guide
5.1 Configuration with Keypad
When the phone is on-hook, press the MENU button to enter MENU mode. When the phone goes offhook or a call comes in, the phone automatically exits the MENU state and prepares for the call.
Here are the Menu options supported:
Menu Functions
Display “Call History”
Press Menu button to enter this menu including
“Received Calls” or
“Dialed Calls” or
“Missed Calls” or
“Back”
Press ‘↓’ or ’↑’ to toggle the selection
Press ‘Å’ to return to the upper menu
Display “Status”
Press Menu button to enter this menu to see the status of the phone
Press ‘↓’ or ’↑’ to toggle the selection
Press Menu or ‘Å’button to exit
Display “Phone Book”
Press Menu button to display the phone book including
“Download Phonebook”
Press ‘↓’ or ’↑’ to toggle the selection
Press Menu button to choose the menu item
Press ‘Å’ button to return to the upper menu
Display “Instant Messages”
Press Menu button to display the Instant Messages received.
Press ‘↓’ or ’↑’ to toggle the selection
Press Menu button to choose the menu item
Press ‘Å’ button to return to the upper menu
Note: GXP2000 only supports the function of receiving of Instant Messages.
Display “Direct IP Call”
Press Menu button to display the direct IP call interface
Enter 12 digit IP address. For example, 10.10.1.2 could be entered like 010010001002.
Press ‘Å’ or ‘Æ’ to move the cursor or toggle the selection
Press Menu button to confirm
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Menu Functions
Display “Preference”
Press Menu button to enter this sub menu including
“Do NOT Disturb” or
“Ring Tone” or
“Ring Volume” or
“Download SCR XML” or
“Erase Custom SCR” or
“Back”
DND (Do NOT Disturb) function could be turned on or off in the “DO NOT Disturb” menu.
Choose different ring tones you prefer in the “Ring Tone” menu.
Adjust ring volume in the “Ring Volume” menu by using ‘Å’ and ‘Æ’ button.
Press ‘↓’ or ’↑’ to toggle the selection
Press Menu button to choose the menu item you want
Press ‘Å’ to return to the upper menu
Display “Configure”
Press Menu button to display the configuration items
“Network” or
“SIP” or
“Audio” or
“Upgrade” or
“Factory Reset”
Please check the web configuration page for more detail information about these items
Press ‘↓’ or ’↑’ to toggle the selection
Press ‘Å’ to return the upper menu
Display “Factory Functions”
Press Menu to display the factory function items including
“Ethernet Loopback” or
“Audio Loopback” or
“Diagnostic Mode” or
“Enable Diag Port” or
“Back”
Press ‘↓’ or ’↑’ to toggle the selection
Press ‘Å’ to return to the upper menu
Display “Reboot”
Press Menu button to reboot the device
Display “Exit”
Press Menu button to exit the menu
25
Menu Functions
Display “Ring Volume”
Press Menu button to hear the selected ring volume, press ‘Å’ or ’Æ’ to hear and adjust the ring
tone volume.
Press Menu button to select and exit, take effect immediately.
Display “Ethernet Loopback”
Press Menu button to enter this mode
A cross Ethernet cable is needed for the test. Before you do the test, plug one end of the cable in
the “PC” port, and the other end in the “LAN” port. You will see the test result on the screen. This
is a feature that is useful for factory as well as for tech team if you need to diagnose if the RJ45
jacks are still good (in terms of hardware).
Note: Running the Ethernet Loopback mode under normal connection will cause IP lost.
Press Menu button to exit the diagnostic mode.
Display “Audio Loopback”
Press Menu button to enter this mode
Tap the keypad to check if the speaker plays the sound caused by your tapping. If yes, the audio
part of your phone works fine. Or you can pick up the handset, and say something to the mic of the
handset. If you can hear what you said from the speaker of the handset, audio part of your phone
works fine.
Press Menu button to exit the mode.
Display “Diagnostic Mode”
Press Menu button to enter this mode, all LEDs will light up
Press any key on the keypad, the button name will be displayed in the LCD. Lift and put back the
handset or press Menu button to exit the diagnostic mode.
Display “Factory Reset”, please be very CAREFUL here
Key in the physical / MAC address on back of the phone, Press Menu button, phone will be reset
back to FACTORY DEFAULT setting, and all your setting will be erased. Please refer to Section
7 for complete details.
26
5.2 Configuration with Web Browser
GXP-2000 series IP phone has an embedded Web server that will respond to HTTP GET/POST
requests. It also has embedded HTML pages that allow a user to configure the IP phone through a Web
browser such as Microsoft’s IE.
5.2.1 Access the Web Configuration Menu
The IP Phone Web Configuration Menu can be accessed by the following URI:
http://Phone-IP-Address
where the Phone-IP-Address is the IP address of the phone.
When the phone is on-hook, press Menu button and then select the Status item to see “IP: IP Address”
NOTE:
•
To type IP address into browser to get into the configuration page, please strip out the leading
“0” as the browser will parse in octet. e.g.: if the IP address is: 192.168.001.014, please type
in: 192.168.1.14.
5.2.2 End User Configuration
Once this HTTP request is entered and sent from a Web browser, the GXP-2000 will respond with the
following login screen:
Grandstream Device Configuration
Password
Login
All Rights Reserved Grandstream Networks, Inc. 2004
The password is case sensitive with maximum length of 25 characters and the factory default password
for End User is “123”.
27
After a correct password is entered in the login screen, the embedded Web server inside the GXP-2000
will respond with the Configuration page which is explained in details below.
Grandstream Device Configuration
STATUS
End User
Password:
IP Address:
BASIC
SETTINGS
ADVANCED
SETTINGS
ACCOUNT ACCOUNT ACCOUNT ACCOUNT
1
2
3
4
(purposely not displayed for security protection)
dynamically assigned via DHCP (default) or PPPoE
(will attempt PPPoE if DHCP fails and following is non-blank)
PPPoE account ID:
PPPoE password:
Preferred DNS server:
0
.
0
statically configured as:
IP Address:
Multi
Purpose
Key 1:
Multi
Purpose
Key 2:
Multi
Purpose
Key 3:
Multi
Purpose
Key 4:
.
0
0
.
192
.
168
.
0
.
160
Subnet Mask:
0
.
0
.
0
.
0
Default Router:
0
.
0
.
0
.
0
DNS Server 1:
0
.
0
.
0
.
0
DNS Server 2:
0
.
0
.
0
.
0
Key Mode:
Name:
Key Mode:
Name:
Key Mode:
Name:
Key Mode:
Name:
Speed Dial
Account:
Account 1
Account:
Account 1
Account:
Account 1
Account:
Account 1
UserID:
Speed Dial
UserID:
Speed Dial
UserID:
Speed Dial
UserID:
28
Multi
Purpose
Key 5:
Multi
Purpose
Key 6:
Multi
Purpose
Key 7:
Time Zone:
Key Mode:
Name:
Key Mode:
Name:
Key Mode:
Name:
Speed Dial
Account:
Account 1
Account:
Account 1
Account:
Account 1
UserID:
Speed Dial
UserID:
Speed Dial
UserID:
GMT-8:00 (US Pacific Time, Los Angeles)
Daylight
No
Yes (if set to Yes, display time will be 1 hour ahead of normal
Savings
4,1,7,2,0;10,-1,7,2,0;60
Time: time) Optional Rule:
LCD
Backlight
No
Yes
Always On:
Time
Display
12 HOUR
24 HOUR
Format:
Date
Display
Format:
Display
Clock
instead of
Date:
Year-Month-Day
Month-Day-Year
Day-Month-Year
No
Yes
System Device Mode
Device
Switch (default)
Mode:
NAT/Router
NAT/Router Configuration
29
WAN side http
No
access: to No)
Yes (WAN side access to http server will be rejected if set
Reply to ICMP
No
on WAN port: No)
Yes (Unit will not respond to PING from WAN side if set to
Cloned WAN
MAC Addr:
LAN Subnet
Mask:
(in hex format)
(default is 255.255.255.0)
(base IP for the LAN port, default is
LAN DHCP
Base IP: 192.168.2.1)
DHCP IP
Lease Time:
120
(in units of hours, default is 120 hours or 5 days)
DMZ IP:
WAN port LAN IP
Port
Forwarding:
LAN port Protocol
0
0
UDP Only
0
0
UDP Only
0
0
UDP Only
0
0
UDP Only
0
0
UDP Only
0
0
UDP Only
0
0
UDP Only
0
0
UDP Only
Update
Cancel
Reboot
All Rights Reserved Grandstream Networks, Inc. 2004, 2005
End User This contains the password to access the Web Configuration Menu.
Password This field is case sensitive with a maximum length of 25 characters.
30
IP Address There are two modes under which the GXP-2000 can operate:
• If DHCP mode is enabled, then all the field values for the
Static IP mode are not used (even though they are still
saved in the Flash memory.) The GXP-2000 will acquire its
IP address from the first DHCP server it discovers from the
LAN it is connected.
• To use the PPPoE feature the PPPoE account settings need
to be set. The GXP-2000 will attempt to establish a PPPoE
session if any of the PPPoE fields is set.
• If Static IP mode is enabled, then the IP address, Subnet
Mask, Default Router IP address, DNS Server 1 (primary),
DNS Server 2 (secondary) fields will need to be configured.
These fields are set to zero by default.
Speed Dial There are 7 speed dial fields that can be configured:
• Name field is used to identify the person. It will be
displayed on LCD when pressing the corresponding key.
• UserID field is the number configured.
• Account field is the SIP account associated with the
number.
Asterisk BLF Asterisk Busy Line Field feature needs the support of Asterisk
PBX. Please check Asterisk for more details.
Presence Watcher This feature is used to monitor the status of other SIP devices
which supports SIP PUBLISH for Presence (RFC 3903). If SIP
PUBLISH for Presence on GXP2000 is enabled (Please refer to
“PUBLISH for Presence” option at account configuration pages),
the status of GXP2000 can be monitored by other SIP devices.
The status of GXP2000 can be changed by enable/disable “Do Not
Disturb” via the GUI Menu. The default status of GXP2000 is “Do
Not Disturb” disabled. The LED of the multi purpose key on the
watcher side will be solid on. Once “Do not disturb” is enable, the
device will send out a PUBLISH message with status update, and
the LED on the Watcher side will be turn off.
Time Zone This parameter controls how the date/time is displayed according
to the specified time zone.
31
Daylight Savings Time This parameter controls whether the displayed time will be
daylight savings time or not. If set to “Yes” and the Optional Rule
is empty, then the displayed time will be 1 hour ahead of normal
time.
The “Automatic Daylight Saving Time Rule” shall have the
following syntax:
start-time;end-time;saving
Both start-time and end-time have the same syntax:
month,day,weekday,hour,minute
month: 1,2,3,..,12 (for Jan, Feb, .., Dec)
day: [+|-]1,2,3,..,31
weekday: 1, 2, 3, .., 7 (for Mon, Tue, .., Sun), or 0 which means the
daylight saving rule is not based on week days but based on the
day of the month.
hour: hour (0-23),
minute: minute (0-59)
If “weekday” is 0, it means the date to start or end daylight saving
is at exactly the given date. In that case, the “day” value must not
be negative. If “weekday” is not zero and “day” is positive, then
the daylight saving starts on the first “day”th iteration of the
weekday (1st Sunday, 3rd Tuesday etc).
If “weekday” is not zero and “day” is negative, then the daylight
saving starts on the last “day”th iteration of the weekday (last
Sunday, 3rd last Tuesday etc).
The saving is in the unit of minutes. The saving time may also be
preceded by a negative (-) sign if subtraction is desired instead of
addition.
The default value for “Automatic Daylight Saving Time Rule”
shall be set to “04,01,7,02,00;10,-1,7,02,00;60” which is the rule for
US.
Examples
US/Canada where daylight saving time is applicable:
04,01,7,02,00;10,-1,7,02,00;60
This means the daylight saving time starts from the first Sunday of
April at 2AM and ends the last Sunday of October at 2AM. The
saving is 60 minutes (1hour).
LCD Backlight Always Allow user to keep the LCD backlight on all the time. Default is
On No.
Time Display Format LCD time display in 12 hour or 24 hour format
32
Date Display Format Allow user to choose among the following three formats:
Year-Month-Day
Month-Day-Year
Day-Month-Year
Display Clock instead of LCD displays clock if set to “Yes”. Default is No.
Date
Device Mode This parameter controls whether the device is working in NAT
router mode or Bridge mode. Need save the setting and reboot the
device before the setting start to work.
WAN side http access If set to “Yes”, user can access the configuration page through the
WAN port, instead of connecting PC and GXP2000 through the
“PC” port to do the configuration. On the other hand, it exposes
the GXP2000 to others, and may cause some security issues for
users. Default is No.
Reply to ICMP on WAN If set to “Yes”, The GXP2000 will respond to the PING command
port from other computers for testing, but it also is vulnerable to the
DOS attack. Default is No.
Cloned WAN MAC Addr Allow the user to set a specific MAC address. Set in Hex format.
LAN Subnet Mask Sets the LAN subnet mask. Default value is 255.255.255.0
LAN DHCP Base IP Base IP for the LAN port, which function as a Gateway for the
subnet.
Default value is 192.168.2.1.
DHCP IP Lease Time Value is set in units of hours. Default value is 120hr (5 Days.) The
time IP address is assigned to the LAN clients.
DMZ IP Forward all WAN IP traffic to a specific IP address if no matching
port is used by GXP-2000 itself or in the defined port forwarding.
Port Forwarding Allow the user to forward a matching (TCP/UDP) port to a specific
LAN IP address with a specific (TCP/UDP) port.
Allow DHCP Option 2 to DHCP Option 2 specifies the offset of the client's subnet in seconds
override Time Zone from Coordinated Universal Time (UTC). The offset is expressed
setting as a two's complement 32-bit integer. A positive offset indicates a
location east of the zero meridian and a negative offset indicates a
location west of the zero meridian. If you choose yes, GXP2000 will
use the time offset resolved from DHCP, instead of the one you
specified in the "Time Zone" option above.
33
In addition to the Basic Settings configuration page, end user also has access to the device Status page.
The following is a screen shot of the device Status page. Details are explained next.
Grandstream Device Configuration
STATUS
BASIC
SETTINGS
ADVANCED
SETTINGS
ACCOUNT ACCOUNT ACCOUNT ACCOUNT
1
2
3
4
00.0B.82.05.11.BC
10.10.1.3
GXP2000
Program-- 1.0.2.6 Bootloader-- 1.0.2.3
0 day(s) 5 hour(s) 56 minute(s)
Account 1: Yes
Account 2: No
Account 3: No
Account 4: Yes
PPPoE Link Up: disabled
detected NAT type is full cone
MAC Address:
IP Address:
Product Model:
Software Version:
System Up Time:
Registered:
All Rights Reserved Grandstream Networks, Inc. 2004, 2005
MAC Address
The device ID, in HEX format. This is a very important ID for ISP
troubleshooting.
IP Address
This field shows LAN IP address of GXP-2000
Product Model
This field contains the product model info.
Software Version
•
•
Program: This is the main software release, its number is always used for
firmware upgrade.
Bootloader: This is normally not changed.
System Up Time
This field shows system up time since the last reboot.
Registered
This field indicates whether the device is registered to the SIP server(s).
PPPoE Link Up
This field shows whether the PPPoE connection is up if connected to DSL
modem.
34
Detected NAT Type
This field shows what kind NAT the GXP-2000 is connected to via its LAN
port. It is based on STUN protocol.
5.2.3 Advanced User Configuration
To login to the Advanced User Configuration page, please follow the instructions in section 5.2.1 to
get to the following login page. The password is case sensitive with a maximum length of 25
characters and the factory default password for Advanced User is “admin”.
Grandstream Device Configuration
Password
Login
All Rights Reserved Grandstream Networks, Inc. 2004
Advanced User configuration includes not only the end user configuration, but also advanced
configuration such as SIP configuration, Codec selection, NAT Traversal Setting and other
miscellaneous configuration. Following is a screen shot of the advanced configuration page:
Grandstream Device Configuration
STATUS
BASIC
SETTINGS
ADVANCED
ACCOUNT 1 ACCOUNT 2 ACCOUNT 3 ACCOUNT 4
SETTINGS
Admin Password:
(purposely not displayed for security
protection)
6.3kbps encoding rate
G723 rate:
No
Silence Suppression:
5.3kbps encoding rate
Yes
2
Voice Frames per TX:
Layer 3 QoS:
(up to 10/20/32/64 for G711/G726/G723/other codecs
respectively)
48
(Diff-Serv or Precedence value)
Layer 2 QoS: 802.1Q/VLAN Tag
7)
35
0
802.1p priority value
0
(0-
No Key Entry Timeout:
Use # as Dial Key:
local RTP port:
4
No
Yes (if set to Yes, "#" will function as the
"(Re-)Dial" key)
5004
Use random port:
keep-alive interval:
Use NAT IP
(in seconds, default is 4 seconds)
(1024-65535, default 5004)
No
20
Yes
(in seconds, default 20 seconds)
(if specified, this will be used in
SIP/SDP message)
(URI or IP:port)
STUN server:
Firmware Upgrade and Upgrade Via
TFTP
Provisioning:
Firmware Server Path:
HTTP
Config Server Path:
Firmware File Prefix:
Firmware File Postfix:
Config File Prefix:
Config File Postfix:
Allow DHCP Option 66 to override server:
No
Yes
Automatic Upgrade:
No
Yes, check for upgrade every
(default 7 days)
10080
minutes
Always Check for New Firmware
Check New Firmware only when F/W pre/suffix changes
Always Skip the Firmware Check
Authenticate Conf File:
No
Yes (cfg file would be authenticated before
acceptance if set to Yes)
Enable Phonebook XML Download:
NO
Phonebook
Download:
XML
YES, HTTP
YES, TFTP
Phonebook XML Server Path:
0
Phonebook Download Interval:
(0-720)
Remove Manually-edited entries on Download:
No
Yes
36
Enable Idle Screen XML Download:
Idle Screen
Download:
XML
NO
YES, HTTP
YES, TFTP
Idle Screen XML Server Path:
DTMF Payload Type:
101
Syslog Server:
Syslog Level:
NTP Server:
NONE
time.nist.gov
(URI or IP address)
Allow DHCP Option 42 to override NTP server:
No
Yes
Custom ring tone 1, used if incoming caller ID is
Distinctive Ring Tone:
Custom ring tone 2, used if incoming caller ID is
Custom ring tone 3, used if incoming caller ID is
Disable Call-Waiting:
No
Yes
Disable Call-Waiting
Tone:
No
Yes
No
Yes
Use Quick IP-call mode:
Lock keypad update:
No
Yes (configuration update via keypad is disabled
if set to Yes)
Update
Cancel
Reboot
All Rights Reserved Grandstream Networks, Inc. 2004, 2005
Admin Administrator password. Only administrator can configure the “Advanced Settings”
Password page. Password field is purposely left blank for security reason after clicking update
and saved. The maximum password length is 25 characters.
G723 rate Encoding rate for G723 codec. By default, 6.3kbps rate is set.
Silence This controls the silence suppression/VAD feature of G723 and G729. If set to
Suppression “Yes”, when a silence is detected, small quantity of VAD packets (instead of audio
packets) will be sent during the period of no talking. If set to “No”, this feature is
disabled.
37
Voice Frames This field contains the number of voice frames to be transmitted in a single packet.
per TX When setting this value, the user should be aware of the requested packet time (used
in SDP message) as a result of configuring this parameter. This parameter is
associated with the first vocoder in the above vocoder Preference List or the actual
used payload type negotiated between the 2 conversation parties at run time.
e.g., if the first vocoder is configured as G723 and the “Voice Frames per TX” is set
to be 2, then the “ptime” value in the SDP message of an INVITE request will be
60ms because each G723 voice frame contains 30ms of audio. Similarly, if this field
is set to be 2 and if the first vocoder chosen is G729 or G711 or G726, then the
“ptime” value in the SDP message of an INVITE request will be 20ms.
If the configured voice frames per TX exceeds the maximum allowed value, the
GXP-2000 will use and save the maximum allowed value for the corresponding first
vocoder choice. The maximum value for PCM is 10(x10ms) frames; for G726, it is
20 (x10ms) frames; for G723, it is 32 (x30ms) frames; for G729/G728, 64 (x10ms)
and 64 (x2.5ms) frames respectively.
Layer 3 QoS This field defines the layer 3 QoS parameter which can be the value used for IP
Precedence or Diff-Serv or MPLS. Default value is 48.
Layer 2 QoS This contains the value used for layer 2 VLAN tag. Default setting is
blank.
No Key Entry Default is 4 seconds.
Timeout
Use # as This parameter allows users to configure the “#” key to be used as the “Send” (or
Send Key “Dial”) key. If set to “Yes”, pressing this key will immediately trigger the sending
of dialed string collected so far. In this case, this key is essentially equivalent to the
“(Re)Dial” key. If set to “No”, this “#” key will then be included as part of the dial
string to be sent out.
Local RTP port This parameter defines the local RTP-RTCP port pair the GXP-2000 will listen and
transmit. It is the base RTP port for channel 0. When configured, channel 0 will use
this port _value for RTP and the port_value+1 for its RTCP; channel 1 will use
port_value+2 for RTP and port_value+3 for its RTCP. The default value is 5004.
Use Random This parameter, when set to Yes, will force random generation of both the local SIP
Port and RTP ports. This is usually necessary when multiple GXP-2000s are behind the
same NAT.
Keep-alive This parameter specifies how often the GXP-2000 sends a blank UDP packet to the
interval SIP server in order to keep the “hole” on the NAT open. Default is 20 seconds.
Use NAT IP NAT IP address used in SIP/SDP message. Default is blank.
STUN Server IP address or Domain name of the STUN server.
38
Firmware This radio button will enable/disable GXP-2000 to download firmware or
Upgrade and configuration file through either TFTP or HTTP.
provisioning
Via TFTP This is the IP address of the configured TFTP server. If selected and it is non-zero
Server or not blank, the GXP-2000 will attempt to retrieve new configuration file or new
code image from the specified TFTP server at boot time. It will make up to 3
attempts before timeout and then it will start the boot process using the existing
code image in the Flash memory. If a TFTP server is configured and a new code
image is retrieved, the new downloaded image will be verified and then saved into
the Flash memory.
Note: Please do NOT interrupt the TFTP upgrade process (especially the power
supply) as this will damage the device. Depending on the network environment this
process can take up to 15 or 20 minutes.
Via HTTP The URL for the HTTP server used for firmware upgrade and configuration via
Server HTTP. For example,
http://provisioning.mycompany.com:6688/Grandstream/1.0.5.16
Here “:6688” is the specific TCP port that the HTTP server is listening at, it can be
omitted if using default port 80.
Note: If Auto Upgrade is set to No, GXP-2000 will only do HTTP download once
at boot up.
Automatic Choose Yes to enable automatic upgrade and provisioning.
Upgrade In “Check for new firmware every” field, enter the number of days to enable GXP2000 to check the server for firmware upgrade or configuration in the defined period
of days.
When set to No, GXP-2000 will only do upgrade once at boot up.
“Always check for New Firmware”
“Check New Firmware only when F/W pre/suffix changes”
“Always Skip the Firmware Check”
Authenticate if set to Yes, cfg file would be authenticated before acceptance. This mechanism is
Conf File useful for the protection of configuration on the device from unauthorized change.
39
Phonebook The feature will be activated when “Enable Downloadable Phonebook” is set to
XML Download YES (HTTP or TFTP) AND a valid “Phonebook XML Path” is set.
When the device boot up and completed the provisioning routine, it will attempt to
download the gs_phonebook.xml file specified in “Phonebook XML Path”. During
this process the LCD should display some messages to indicate that XML
Phonebook download is in progress.
If the “Phonebook Download Interval” is set to a non-zero value, GXP2000 will
periodically check and download the updated phonebook available.
If the “Remove manually edited entries on download” option is set to NO (by
default), GXP2000 will keep ALL entries edited manually, and insert the
downloaded entries, and then save the phonebook. Otherwise ALL the existing
phonebook entries on the phone will be erased and filled with the downloaded
entries.
At any time, you can trigger an immediate download from by choosing the
“Download Phonebook” in the GUI Phone Book Menu (you can use the down
arrow key when the phone is on-hook).
Example XML file of gs_phonebook.xml:
<?xml version="1.0"?>
<AddressBook>
<Contact>
<LastName>Craig</LastName>
<FirstName>Richard</FirstName>
<Phone>
<phonenumber>6910</phonenumber>
<accountindex>0</accountindex>
</Phone>
</Contact>
<Contact>
<LastName>Peterson</LastName>
<FirstName>Susan</FirstName>
<Phone>
<phonenumber>6911</phonenumber>
<accountindex>0</accountindex>
</Phone>
</Contact>
<Contact>
<LastName>Mayer</LastName>
<FirstName>Cindy</FirstName>
<Phone>
<phonenumber>6915</phonenumber>
<accountindex>0</accountindex>
</Phone>
</Contact>
</AddressBook>
Note: This feature is designed for enterprise user only. Please contact Grandstream
for more details.
40
Idle Screen The feature will be activated when “Enable Idle-Screen XML Download” is set to
XML Download YES (HTTP or TFTP) AND a valid “Idle-Screen XML Path” is set.
This feature does not automatically download the gs_screen.xml file in the path
even when activated. Because the LCD is composed of 130*64 mono pixels, the
resolution of the screen XML should be within this range. The following 2 options
are added to the Preference LCD GUI submenu:
Download SCR XML
Erase Custom SCR
User will have to choose to Download SCR XML to start the download process.
Once the XML is successfully downloaded it will be effective right away. The file
will be saved and loaded automatically after reboot.
Example XML file of gs_screen.xml
<?xml version="1.0"?>
<Screen>
<IdleScreen>
<ShowStatusLine>false</ShowStatusLine>
<DisplayBitmap>
<Bitmap>Put your customized screen file with bitmap format here</Bitmap>
<X>0</X>
<Y>0</Y>
</DisplayBitmap>
<DisplayString font="f8" halign="Right">
<DisplayStr>Doraemon</DisplayStr>
<X>130</X>
<Y>0</Y>
</DisplayString>
<DisplayString font="f10" halign="Left" valign="Bottom">
<DisplayStr>Call me:</DisplayStr>
<X>0</X>
<Y>54</Y>
</DisplayString>
<DisplayString font="f8" halign="Left" valign="Bottom">
<DisplayStr>$X@$V</DisplayStr>
<X>0</X>
<Y>64</Y>
</DisplayString>
</IdleScreen>
</Screen>
Note: The feature requires some expertise on XML. For more technical details,
please contact Grandstream.
DTMF Payload This parameter sets the payload type for DTMF using RFC2833.
Type
NTP server This parameter defines the URI or IP address of the NTP (Network Time Protocol)
server which is used by GXP-2000 to display the current date/time.
Distinctive Ring Customer Ring Tone 1 to 3 with associate Caller ID: when selected, if Caller ID is
Tone configured, then the device will ONLY sound this ring tone when the incoming call
is from the Caller ID, device will use System Ring Tone for all other calls.
When selected but no Caller ID is configured, the selected ring tone will be used for
all incoming calls.
Disable Call Default is No.
Waiting
41
Disable Call- Default is No. This feature is dedicated for busy sales/service call centers. The call
Waiting Tone waiting sound is turned off, but LINE LED flashing still can be seen, and can take
the call by pressing the LINE button.
Quick IP-call Please refer user manual chapter 4.4.4.
mode
Lock keypad If this parameter is set to “Yes”, the configuration updates via keypad for Menu
update Item 7, 9, 12 are disabled.
Syslog Server The IP address or URL of System log server. This feature is especially useful for
ITSP (Internet Telephone Service Provider)
Syslog Level Select the ATA to report the log level. Default is NONE. The level is one of
DEBUG, INFO, WARNING or ERROR. Syslog messages are sent based on the
following events:
•
•
•
•
•
•
•
•
•
•
product model/version on boot up (INFO level)
NAT related info (INFO level)
sent or received SIP message (DEBUG level)
SIP message summary (INFO level)
inbound and outbound calls (INFO level)
registration status change (INFO level)
negotiated codec (INFO level)
Ethernet link up (INFO level)
SLIC chip exception (WARNING and ERROR levels)
memory exception (ERROR level)
The Syslog uses USER facility. In addition to standard Syslog payload, it contains
the following components:
GS_LOG: [device MAC address][error code] error message
Here is an example: May 19 02:40:38 192.168.1.14 GS_LOG:
[00:0b:82:00:a1:be][000] Ethernet link is up
Allow DHCP DHCP Option 66 is used to identify a TFTP server when the 'sname' field in the
Option 66 to DHCP header has been used for DHCP options. If you choose yes, GXP2000 will
override server
use the TFTP server resolved from DHCP, instead of the one you specified in the
"TFTP Server" option above.
Allow DHCP DHCP Option 42 specifies a list of IP addresses for Network Time Protocol (NTP)
Option 42 to servers available to the client. If you choose yes, GXP2000 will use the NTP servers
override NTP
resolved from DHCP, instead of the one you specified in the "NTP Server" option
server
above.
42
Four independent SIP accounts each has its own configuration page.
identical. The following is a screen shot of SIP Account 1 settings.
Their configurations are
Grandstream Device Configuration
STATUS
BASIC
SETTINGS
ADVANCED
SETTINGS
Account Active:
No
Account Name:
MyCompany
SIP Server:
ACCOUNT ACCOUNT ACCOUNT ACCOUNT
1
2
3
4
Yes
(e.g., MyCompany)
sip.mycompany.com
address)
(e.g., proxy.myprovider.com, or IP address, if
Outbound Proxy: any)
SIP User ID:
Authenticate ID:
Authenticate Password:
(e.g., sip.mycompany.com, or IP
(the user part of an SIP address)
(can be identical to or different from SIP User
ID)
(purposely not displayed for security
protection)
Name:
(optional, e.g., John Doe)
Use DNS SRV:
No
Yes
User ID is phone number:
No
Yes
SIP Registration:
No
Yes
Unregister On Reboot:
No
Yes
60
(in minutes. default 1 hour, max 45 days)
local SIP port:
5060
(default 5060)
SIP T1 Timeout:
1 sec
SIP T2 Interval:
4 sec
Register Expiration:
NAT Traversal (STUN):
No
No, but send keep-alive
PUBLISH for Presence:
No
Yes
43
Yes
SUBSCRIBE for MWI:
No
Yes
Proxy-Require:
Voice Mail UserID:
Send DTMF:
(User ID/extension for 3rd party voice mail
system)
in-audio
INFO
via RTP (RFC2833)
No
Early Dial: response)
No
Yes (if Yes, Call Forwarding & CallWaiting-Disable are supported locally)
No
Disable Missed-Call:
Session Expiration:
Min-SE:
Yes (use "Yes" only if proxy supports 484
(this prefix string is added to each dialed number)
Dial Plan Prefix:
Enable Call Features:
Yes (Missed calls NOT recorded)
180
(in seconds. default 180 seconds)
90
(in seconds. default and minimum 90 seconds)
Caller Request Timer:
No
outbound calls)
Callee Request Timer:
No
request one)
Force Timer:
via SIP
Yes (Request for timer when making
Yes (When caller supports timer but did not
No
Yes (Use timer even when remote party
does not support)
UAC Specify Refresher:
UAC
UAS
UAS Specify Refresher:
UAC
UAS (When UAC did not specify refresher
tag)
Force INVITE:
No
of UPDATE)
Enable 100rel:
No
Yes (Always refresh with INVITE instead
Yes
system ring tone
Account Ring Tone:
Omit (Recommended)
custom ring tone 1
custom ring tone 2
44
custom ring tone 3
Send Anonymous:
No
Yes (caller ID will be blocked if set to Yes)
Auto Answer:
No
Yes
No
Yes
No
Yes
No
Yes
Allow Auto Answer by CallInfo:
Turn off speaker on remote
disconnect:
Check SIP User ID for
incoming INVITE:
Preferred Vocoder:
(in listed order)
SRTP Mode:
choice 1:
PCMU
choice 5:
PCMU
choice 2:
PCMA
choice 6:
PCMU
choice 3:
G.723.1
choice 7:
PCMU
choice 4:
G.729A/B
choice 8:
PCMU
Disabled
forced
Special Feature:
Enabled but not forced
Enabled and
Standard
Update
Cancel
Reboot
All Rights Reserved Grandstream Networks, Inc. 2004, 2005
Individual Account Settings
Account Active
This field indicates whether the account is active or not. The default value for
the primary account Account 1 is Yes. The default values for the other three
accounts are No.
Account Name
A name to identify an account which will be displayed in LCD.
SIP Server
SIP Server’s IP address or Domain name provided by VoIP service provider.
Outbound Proxy
IP address or Domain name of Outbound Proxy, or Media Gateway, or
Session Border Controller. Used by GXP-2000 for firewall or NAT
penetration in different network environment. If symmetric NAT is detected,
STUN will not work and ONLY outbound proxy can provide solution for it.
45
SIP User ID
User account information, provided by VoIP service provider (ITSP), usually
has the form of digit similar to phone number or actually a phone number.
Authenticate ID
SIP service subscriber’s Authenticate ID used for authentication. Can be
identical to or different from SIP User ID.
Authenticate
Password
SIP service subscriber’s account password for GXP-2000 to register to (SIP)
servers of ITSP.
Name
SIP service subscriber’s name which will be used for Caller ID display.
Use DNS SRV:
Default is No. If set to Yes the client will use DNS SRV to look up server.
User ID is Phone
Number
If the GXP-2000 has an assigned PSTN telephone number, this field should
be set to “Yes”. Otherwise, set it to “No”. If “Yes” is set, a “user=phone”
parameter will be attached to the “From” header in SIP request
SIP Registration
This parameter controls whether the GXP-2000 needs to send REGISTER
messages to the proxy server. The default setting is “Yes”.
Unregister on
Reboot
Default is No. If set to yes, the SIP user’s registration information will be
cleared on reboot.
Register Expiration This parameter allows user to specify the time frequency (in minutes) that
GXP-2000 refreshes its registration with the specified registrar. The default
interval is 60 minutes (or 1 hour). The maximum interval is 65535 minutes
(about 45 days).
Local SIP port
This parameter defines the local SIP port the GXP-2000 will listen and
transmit. The default value for Account 1 is 5060. It is 5062, 5064, 5066 for
Account 2, Account 3 and Account 4 respectively.
SIP T1 Timeout
T1 is an estimate of the round-trip time (RTT) between the client and server
transactions. If the network latency is high, select bigger value for reliable
usage.
SIP T2 Interval
This element sets the value of the SIP protocol T2 timer, in seconds. Timer
T2 defines the retransmit interval for INVITE responses and non-INVITE
requests. The SIP protocol default value is 4 seconds.
46
NAT Traversal
This parameter defines whether the GXP-2000 NAT traversal mechanism
will be activated or not. If activated (by choosing “Yes”) and a STUN server
is also specified, then the GXP-2000 will behave according to the STUN
client specification. Under this mode, the embedded STUN client inside the
GXP-2000 will attempt to detect if and what type of firewall/NAT it is sitting
behind through communication with the specified STUN server. If the
detected NAT is a Full Cone, Restricted Cone, or a Port-Restricted Cone, the
GXP-2000 will attempt to use its mapped public IP address and port in all of
its SIP and SDP messages. If the NAT Traversal field is set to “Yes” with no
specified STUN server, the GXP-2000 will periodically (every 20 seconds or
so) send a blank UDP packet (with no payload data) to the SIP server to keep
the “hole” on the NAT open.
Subscribe for
MWI:
Default is No. When set to “Yes” a SUBSCRIBE for Message Waiting
Indication will be sent periodically.
PUBLISH for
Presence:
GXP2000 supports SIP PUBLISH for Presence (RFC 3903). If SIP
PUBLISH for Presence on GXP2000 is enabled, the status of GXP2000 can
be monitored by other SIP devices (Please refer to “Presence Watcher”
function on Multi Purpose Keys).
The status of GXP2000 can be toggled by enabling/disabling “Do Not
Disturb” via the GUI Menu. The default status of GXP2000 is “Do Not
Disturb” disabled. The LED of the multi purpose key on the watcher side will
be solid on. Once “Do not disturb” is enable, the device will send out a
PUBLISH message with status update, and the LED on the Watcher side will
be turn off.
Proxy-Require
SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.
Voice Mail User ID
When configured, user will be able to dial voice mail server by pressing
“MSG” button.
Send DTMF
This parameter specifies the mechanism to transmit DTMF digit. There are 3
modes supported: in audio which means DTMF is combined in audio signal
(not very reliable with low-bit-rate codec), via RTP (RFC2833), or via SIP
INFO.
Early Dial
Default is No. Use only if proxy supports 484 response.
Dial Plan Prefix
Sets the prefix added to each dialed number.
Enable Call
Features
Default is No. If set to Yes, Call transfer, Call Forwarding & Do-Not-Disturb
are supported locally.
Disable Missed-Call Default is No. If set to Yes, missed calls will not be recorded for your review.
47
Session Expiration
Grandstream implemented SIP Session Timer. The session timer extension
enables SIP sessions to be periodically “refreshed” via a SIP request
(UPDATE, or re-INVITE. Once the session interval expires, if there is no
refresh via a UPDATE or re-INVITE message, the session will be terminated.
Session Expiration is the time (in seconds) at which the session is considered
timed out, if no successful session refresh transaction occurs beforehand. The
default value is 180 seconds.
The minimum session expiration (in seconds).
seconds.
Min-SE
The default value is 90
Caller
Timer
Request If selecting “Yes” the phone will use session timer when it makes outbound
calls if remote party supports session timer.
Callee
Timer
Request If selecting “Yes” the phone will use session timer when it receives inbound
calls with session timer request.
If selecting “Yes” the phone will use session timer even if the remote party
does not support this feature. Selecting “No” will allow the phone to enable
session timer only when the remote party support this feature.
To turn off Session Timer, select “No” for Caller Request Timer, Callee
Request Timer, and Force Timer.
Force Timer
UAC
Refresher
Specify As a Caller, select UAC to use the phone as the refresher, or UAS to use the
Callee or proxy server as the refresher.
UAS
Refresher
Specify As a Callee, select UAC to use caller or proxy server as the refresher, or UAS
to use the phone as the refresher.
Force INVITE
Session Timer can be refreshed using INVITE method or UPDATE method.
Select “Yes” to use INVITE method to refresh the session timer.
Enable 100rel
The use of the PRACK (Provisional Acknowledgment) method enables
reliability to be offered to SIP provisional responses (1xx series). This is very
important if PSTN internetworking is to be supported. A user’s wish to use
reliable provisional responses is invoked by the 100rel tag which is appended
to the value of the required header of initial signalling messages.
Account Ring Tone There are 4 different ring tone that are defined:
• System Ring Tone: when selected, all calls will ring with system ring
tone.
• Customer Ring Tone 1 to 3: when selected, GXP-2000 will ONLY
play this ring tone for all the incoming calls for this account.
Send Anonymous
If this parameter is set to “Yes”, the “From” header in outgoing INVITE
message will be set to anonymous, essentially blocking the Caller ID from
displaying.
48
Auto Answer
When set to “Yes”, GXP-2000 will automatically switch to speaker when
there is an incoming call.
Allow Auto Answer Default is No. If set to Yes, auto answer depends on the Call-Info in the SIP
message. This feature needs the support of IP-PBX.
by Call-Info
Turn off speaker on Default is No. If set to Yes, the speaker will turn off, and the phone will go
back to idle status, after the other party of the call hands up.
remote disconnect
Check SIP User ID
for incoming
INVITE
Default is No. If set to Yes, this account will deny direct IP call, and the call
to previous configured account will be refused also.
Preferred Vocoder
The GXP-2000 supports up to 5 different Vocoder types including G.711 A/U-law, GSM, G.723.1, G.729A/B.
User can configure Vocoders in a preference list that will be included with
the same preference order in SDP message. The first Vocoder in this list can
be entered by choosing the appropriate option in “Choice 1”. Similarly, the
last Vocoder in this list can be entered by choosing the appropriate option in
“Choice 8”.
SRTP Mode
•
•
•
Special Feature
Disabled: GXP2000 will use RTP for both inbound and outbound
calls. (Default setting)
Enabled but not forced: GXP2000 provides crypto suites to others SIP
phones. If they support SRTP, then use SRTP, otherwise, use RTP.
Enabled and forced: All calls using this account must use SRTP. If the
other party doesn’t support SRTP, GXP2000 will decline the call. If
we invite others, and receive a “200 Ok” message without crypto
suite, we will send a “BYE” message followed by the “ACK”
message to terminate the call. If someone invites us without crypto
suite, we will response a “488 Not Acceptable Here” message.
Default is Standard. Choose the selection to meet some special requirements
from Soft Switch vendors like Nortel, Broadsoft, etc.
5.2.4 Saving the Configuration Changes
Once a change is made, the user should press the “Update” button in the Configuration Menu. The IP
phone will then display the following screen to confirm that the changes have been saved:
49
Grandstream Device Configuration
STATUS
BASIC SETTINGS
ADVANCED SETTINGS
ACCOUNT 1
ACCOUNT 2
ACCOUNT 3
ACCOUNT 4
Your configuration changes have been saved.
They will take effect on next reboot.
All Rights Reserved Grandstream Networks, Inc. 2004
User is recommended to power cycle the IP phone after seeing the above message.
5.2.5
Rebooting the Phone from Remote
The administrator of the phone can remotely reboot the phone by pressing the “Reboot” button at the
bottom of the configuration menu. Once done, the following screen will be displayed to indicate that
rebooting is underway.
Grandstream Device Configuration
The device is rebooting now...
You may relogin by clicking on the link below in 30 seconds.
Click to relogin
All Rights Reserved Grandstream Networks, Inc. 2004
At this point, user can relogin to the phone after waiting for about 30 seconds.
5.3 Configuration through Central Provisioning Server
50
Grandstream GXP-2000 can be automatically configured from a central provisioning system.
When GXP-2000 boots up, it will send TFTP or HTTP request to download configuration file
“cfg000b82xxxxxx”, where “000b82xxxxxx” is the MAC address of the GXP-2000.
The configuration files can be downloaded via TFTP or HTTP from the central server. A service
provider or an enterprise with large deployment of GXP-2000 can easily manage the configuration and
service provisioning of individual devices remotely from a central server.
Grandstream provides a licensed provisioning system that can be used to support automated
configuration of GXP-2000. It uses enhanced (NAT friendly) TFTP or HTTP (thus no NAT issues)
and other communication protocols to communicate with each individual GXP-2000 for firmware
upgrade, remote reboot, etc.
Grandstream provide redirection service to VoIP service providers. It could be either simple
redirection or with certain special provisioning settings. Initially upon booting up, Grandstream
devices by default point to Grandstream provisioning server, based on the unique MAC address of
each device, our redirection service provision the devices with redirection settings so that they will be
redirected to customer’s TFTP or http server for further provisioning. Grandstream also provide
GAPSLite software package which contains our NAT friendly TFTP server and a configuration tool to
facilitate the task of generating device configuration files.
The GAPSLite configuration tool is now free to end users. The tool and configuration templates can be
downloaded from
http://www.grandstream.com/DOWNLOAD/Configuration_Tool/.
For details on how redirection service works, please refer to the documentation of provisioning
product.
51
6 Firmware Upgrade
Software upgrade can be done via either TFTP or HTTP. The corresponding configuration settings are
in the ADVANCED SETTINGS configuration page.
6.1 Firmware Upgrade through TFTP/HTTP
To upgrade via TFTP or HTTP, the “Firmware Upgrade and Provisioning upgrade via” field needs to
be set to TFTP or HTTP, respectively. “Firmware Server Path” needs to be set to a valid URL of a
TFTP or HTTP server, server name can be in either FQDN or IP address format. Here are examples of
some valid URL.
e.g. firmware.mycompany.com:6688/Grandstream/1.1.1.14
e.g. 168.75.215.189
NOTES:
• TFTP server in IP address format can be configured via GUI menu or web configuration
page. If TFTP server is in FQDN format, it must be set via web configuration interface.
• Once a “Firmware Server Path” is set, user needs to update the settings and reboot the
device. If the configured firmware server is found and a new code image is available, the
GXP2000 will attempt to retrieve the new image files by downloading them into the
GXP2000’s SRAM. Upon verification of checksum, the new code image will then be saved
into the Flash. During the firmware upgrading process, the GXP2000’s LCD will display
the message about the firmware loading progress. If TFTP/HTTP fails for any reason (e.g.,
TFTP/HTTP server is not responding, there are no code image files available for upgrade,
or checksum test fails, etc), the GXP2000 will stop the TFTP/HTTP process and simply
boot using the existing code image in the flash.
• Firmware upgrade may take as long as 5 to 20 minutes over Internet, or just 2 minutes if it
is performed on a LAN. It is recommended to conduct firmware upgrade in a controlled
LAN environment if possible. For users who do not have a local firmware upgrade server,
Grandstream provides a NAT-friendly TFTP server on the public Internet for firmware
upgrade. Please check the Services section of Grandstream’s Web site at
http://www.grandstream.com/y-firmware.htm to obtain our public TFTP server’s IP
address.
• Alternatively, user can download a free TFTP or HTTP server and conduct local firmware
upgrade. A free windows version TFTP server is available for download from
http://support.solarwinds.net/updates/New-customerFree.cfm. Our latest official release can
be downloaded from http://www.grandstream.com/y-firmware.htm. Unzip the file and put
all of them under the root directory of the TFTP server. Put the PC running the TFTP server
and the GXP2000 in the same LAN segment. Please go to File -> Configure -> Security to
change the TFTP server's default setting from "Receive Only" to "Transmit Only" for the
firmware upgrade. Start the TFTP server, in the GXP2000’s web configuration page,
configure the Firmware Server Path with the IP address of the PC, update the change and
reboot the unit.
52
6.2 Configuration File Download
Grandstream SIP Device can be configured via Web Interface as well as via Configuration File
through TFTP or HTTP. “Config Server Path” is the TFTP or HTTP server path for configuration file.
It needs to be set to a valid URL, either in FQDN or IP address format. The “Config Server Path” can
be same or different from the “Firmware Server Path”.
A configuration parameter is associated with each particular field in the web configuration page. A
parameter consists of a Capital letter P and 2 to 3 (Could be extended to 4 in the future) digit numeric
numbers. i.e., P2 is associated with “Admin Password” in the ADVANCED SETTINGS page. For a
detailed parameter list, please refer to the corresponding firmware release configuration template.
When Grandstream Device boots up or reboots, it will issue request for configuration file named
“cfgxxxxxxxxxxxx”, where “xxxxxxxxxxxx” is the MAC address of the device, i.e.,
“cfg000b820102ab”. The configuration file name should be in lower cases.
6.3 Firmware and Configuration File Prefix and Postfix
The firmware release 1.1.1.14 allow user to customize the firmware file name by adding prefix and
postfix for both firmware and configuration file.
Firmware Prefix and Postfix allows device to download the firmware name with the matching Prefix
and Postfix. This makes it the possible to store ALL of the firmware with different version in one
single directory. Similarly, Config File Prefix and Postfix allows device to download the configuration
file with the matching Prefix and Postfix. Thus multiple configuration files for the same device can be
stored in one directory.
In addition, when the field “Check New Firmware only when F/W pre/suffix changes” is set to “Yes”,
the device will only issue firmware upgrade request if there are changes in the firmware Prefix or
Postfix.
6.4 Managing Firmware and Configuration File Download
When “Automatic Upgrade” is set to “Yes”, Service Provider can use P193 (Auto Check Interval, in
minutes, default and minimum is 60 minutes) to have the devices periodically check with either
Firmware Server or Config Server, whenever they are defined. This allows the device periodically
check if there are any new changes need to be taken on a scheduled time. By defining different
intervals in P193 for different devices, Server Provider can spread the Firmware or Configuration File
download in minutes to reduce the Firmware or Provisioning Server load at any given time.
53
7 Restore Factory Default Setting
Warning !!!
Restore the Factory Default Setting will DELETE all configuration information of the device.
Please backup or print out all the settings before you approach to following steps. Grandstream will
not take any responsibility if you lose all the parameters of setting and cannot connect to your
service provider.
Disconnect network cable and power cycle the unit before resetting factory default settings.
1. Step 1: Press round menu key to bring up the key pad configuration UI menu, select “Config”, press
“OK” to enter submenu, select “Factory Reset”
2. Step 2: Key in the MAC address printed on the bottom of the sticker on the back. Please use the
following mapping:
0-9: 0-9
a. A: 22 (press the “2” key twice, “A” will show on the LCD)
b. B: 222
c. C: 2222
d. D: 33 (press the “3” key twice, “D” will show on the LCD)
e. E: 333
f. F: 3333
For example: if the MAC address is 000b8200e395, it should be key in as “0002228200333395”.
NOTE: If there are digits like “22” in the MAC, you need to type “2” then press “->” right arrow key to move
the cursor or wait for 4 seconds to continue to key in another “2”.
3. Step 3: Press the round menu key again to move the cursor to “OK” button. Press round menu key
again to confirm. If the MAC address is correct, the phone will reboot. Otherwise, it will exit to previous
keypad menu interface.
54
Appendix I
Glossary of Terms
ADSL
Asymmetric Digital Subscriber Line: Modems attached to twisted pair copper wiring that
transmit from 1.5 Mbps to 9 Mbps downstream (to the subscriber) and from 16 kbps to 800
kbps upstream, depending on line distance.
AGC
Automatic Gain Control, is an electronic system found in many types of devices. Its purpose is
to control the gain of a system in order to maintain some measure of performance over a
changing range of real world conditions.
ARP
Address Resolution Protocol is a protocol used by the Internet Protocol (IP) [RFC826],
pecifically IPv4, to map IP network addresses to the hardware addresses used by a data link
protocol. The protocol operates below the network layer as a part of the interface between the
OSI network and OSI link layer. It is used when IPv4 is used over Ethernet
ATA
Analogue Telephone Adapter. Covert analogue telephone to be used in data network for VoIP,
like Grandstream HT series products.
CODEC
Abbreviation for Coder-Decoder. It's an analog-to-digital (A/D) and digital-to-analog (D/A)
converter for translating the signals from the outside world to digital, and back again.
CNG
Comfort Noise Generator, geneate artificial background noise used in radio and wireless
communications to fill the silent time in a transmission resulting from voice activity detection.
DATAGRAM
A data packet carrying its own address information so it can be independently routed from its
source to the destination computer
DECIMATE
To discard portions of a signal in order to reduce the amount of information to be encoded or
compressed. Lossy compression algorithms ordinarily decimate while subsampling.
DECT
Digital Enhanced Cordless Telecommunications: A standard developed by the European
Telecommunication Standard Institute from 1988, governing pan-European digital mobile
telephony. DECT covers wireless PBXs, telepoint, residential cordless telephones, wireless
access to the public switched telephone network, Closed User Groups (CUGs), Local Area
Networks, and wireless local loop. The DECT Common Interface radio standard is a
multicarrier time division multiple access, time division duplex (MC-TDMA-TDD) radio
55
transmission technique using ten radio frequency channels from 1880 to 1930 MHz, each
divided into 24 time slots of 10ms, and twelve full-duplex accesses per carrier, for a total of
120 possible combinations. A DECT base station (an RFP, Radio Fixed Part) can transmit all
12 possible accesses (time slots) simultaneously by using different frequencies or using only
one frequency. All signaling information is transmitted from the RFP within a multiframe (16
frames). Voice signals are digitally encoded into a 32 kbit/s signal using Adaptive Differential
Pulse Code Modulation.
DNS
Short for Domain Name System (or Service or Server), an Internet service that translates
domain names into IP addresses
DID
Direct Inward Dialing
Direct Inward Dialing. The ability for an outside caller to dial to a PBX extension without
going through an attendant or auto-attendant.
DSP
Digital Signal Processing. Using computers to process signals such as sound, video, and other
analog signals which have been converted to digital form.
Digital Signal Processor. A specialized CPU used for digital signal processing.
Grandstream products all have DSP chips built inside.
DTMF
Dual Tone Multi Frequency
The standard tone-pairs used on telephone terminals for dialing using in-band signaling. The
standards define 16 tone-pairs (0-9, #, * and A-F) although most terminals support only 12 of
them (0-9, * and #).
FQDN
Fully Qualified Domain Name
A FQDN consists of a host and domain name, including top-level domain. For example,
www.grandstream.com is a fully qualified domain name. www is the host, grandstream is the
second-level domain, and.com is the top level domain.
FXO
56
Foreign eXchange Office
An FXO device can be an analog phone, answering machine, fax, or anything that handles a
call from the telephone company like AT&T. They should also operate the same way when
connected to an FXS interface.
An FXO interface will accept calls from FXS or PSTN interfaces. All countries and regions
have their own standards.
FXO is complimentary to FXS (and the PSTN).
FXS
Foreign eXchange Station
An FXS device has hardware to generate the ring signal to the FXO extension (usually an
analog phone).
An FXS device will allow any FXO device to operate as if it were connected to the phone
company. This makes your PBX the POTS+PSTN for the phone.
The FXS Interface connects to FXO devices (by an FXO interface, of course).
DHCP
The Dynamic Host Configuration Protocol (DHCP) is an Internet protocol for automating the
configuration of computers that use TCP/IP. DHCP can be used to automatically assign IP
addresses, to deliver TCP/IP stack configuration parameters such as the subnet mask and
default router, and to provide other configuration information such as the addresses for printer,
time and news servers.
ECHO CANCELLATION
Echo Cancellation is used in telephony to describe the process of removing echo from a voice
communication in order to improve voice quality on a telephone call. In addition to improving
quality, this process improves bandwidth savings achieved through silence suppression by
preventing echo from traveling across a network.
There are two types of echo of relevance in telephony: acoustic echo and hybrid echo. Speech
compression techniques and digital processing delay often contribute to echo generation in
telephone networks.
H.323
A suite of standards for multimedia conferences on traditional packet-switched networks.
HTTP
57
Hyper Text Transfer Protocol; the World Wide Web protocol that performs the request and
retrieve functions of a server
IP
Internet Protocol. A packet-based protocol for delivering data across networks.
IP-PBX
IP-based Private Branch Exchange
IP Telephony
(Internet Protocol telephony, also known as Voice over IP Telephony) A general term for the
technologies that use the Internet Protocol's packet-switched connections to exchange voice,
fax, and other forms of information that have traditionally been carried over the dedicated
circuit-switched connections of the public switched telephone network (PSTN). The basic steps
involved in originating an IP Telephony call are conversion of the analog voice signal to digital
format and compression/translation of the signal into Internet protocol (IP) packets for
transmission over the Internet or other packet-switched networks; the process is reversed at the
receiving end. The terms IP Telephony and Internet Telephony are often used to mean the
same; however, they are not 100 per cent interchangeable, since Internet is only a subcase of
packet-switched networks. For users who have free or fixed-price Internet access, IP Telephony
software essentially provides free telephone calls anywhere in the world. However, the
challenge of IP Telephony is maintaining the quality of service expected by subscribers.
Session border controllers resolve this issue by providing quality assurance comparable to
legacy telephone systems.
IVR
IVR is a software application that accepts a combination of voice telephone input and touchtone keypad selection and provides appropriate responses in the form of voice, fax, callback, email and perhaps other media.
MTU
A Maximum Transmission Unit (MTU) is the largest size packet or frame, specified in octets
(eight-bit bytes), that can be sent in a packet- or frame-based network such as the Internet. The
maximum for Ethernet is 1500 byte.
NAT
Network Address Translation
NTP
Network Time Protocol, a protocol to exchange and synchronize time over networks
The port used is UDP 123
Grandstream products using NTP to get time from Internet
OBP/SBC
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Outbound Proxy or another name Session Border Controller. A device used in VoIP networks.
OBP/SBCs are put into the signaling and media path between calling and called party. The
OBP/SBC acts as if it was the called VoIP phone and places a second call to the called party.
The effect of this behaviour is that not only the signaling traffic, but also the media traffic
(voice, video etc) crosses the OBP/SBC. Without an OBP/SBC, the media traffic travels
directly between the VoIP phones. Private OBP/SBCs are used along with firewalls to enable
VoIP calls to and from a protected enterprise network. Public VoIP service providers use
OBP/SBCs to allow the use of VoIP protocols from private networks with internet connections
using NAT.
PPPoE
Point-to-Point Protocol over Ethernet, is a network protocol for encapsulating PPP frames in
Ethernet frames. It is used mainly with cable modem and DSL services.
PSTN
Public Switched Telephone Network
i.e. the phone service we use for every ordinary phone call, or called POT (Plain Old
Telephone), or circuit switched network.
RTCP
Real-time Transport Control Protocol, defined in RFC 3550, a sister protocol of the Real-time
Transport Protocol (RTP), It partners RTP in the delivery and packaging of multimedia data,
but does not transport any data itself. It is used periodically to transmit control packets to
participants in a streaming multimedia session. The primary function of RTCP is to provide
feedback on the quality of service being provided by RTP.
RTP
Real-time Transport Protocol defines a standardized packet format for delivering audio and
video over the Internet. It was developed by the Audio-Video Transport Working Group of the
IETF and first published in 1996 as RFC 1889
SDP
Session Description Protocol, is a format for describing streaming media initialization
parameters. It has been published by the IETF as RFC 2327.
SIP
Session Initiation Protocol, An IP telephony signaling protocol developed by the IETF
(RFC3261). SIP is a text-based protocol suitable for integrated voice-data applications. SIP is
designed for voice transmission and uses fewer resources and is considerably less complex than
H.323.
All Grandstream products are SIP based
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STUN
Simple Traversal of UDP over NATs, is a network protocol allowing clients behind NAT
(or multiple NATs) to find out its public address, the type of NAT it is behind and the internet
side port associated by the NAT with a particular local port. This information is used to set up
UDP communication between two hosts that are both behind NAT routers. The protocol is
defined in RFC 3489. STUN will usually work good with non-symmetric NAT routers.
TCP
Transmission Control Protocol, is one of the core protocols of the Internet protocol suite.
Using TCP, applications on networked hosts can create connections to one another, over which
they can exchange data or packets. The protocol guarantees reliable and in-order delivery of
sender to receiver data.
TFTP
Trivial File Transfer Protocol, is a very simple file transfer protocol, with the functionality of
a very basic form of FTP; It uses UDP (port 69) as its transport protocol.
UDP
User Datagram Protocol (UDP) is one of the core protocols of the Internet protocol suite.
Using UDP, programs on networked computers can send short messages known as datagrams
to one another. UDP does not provide the reliability and ordering guarantees that TCP does;
datagrams may arrive out of order or go missing without notice. However, as a result, UDP is
faster and more efficient for many lightweight or time-sensitive purposes.
VAD
Voice Activity Detection or Voice Activity Detector is an algorithm used in speech
processing wherein, the presence or absence of human speech is detected from the audio
samples.
VLAN
A virtual LAN, known as a VLAN, is a logically-independent network. Several VLANs can coexist on a single physical switch. It is usually refer to the IEEE 802.1Q tagging protocol.
VoIP
Voice over IP
VoIP encompasses many protocols. All the protocols do some form of signalling of call
capabilities and transport of voice data from one point to another. e.g: SIP, H.323, etc.
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Appendix II GUI Menu Chart
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